mdw@git.distorted.org.uk Git - disorder/blob - clients/playrtp.c

   1 /*
   2  * This file is part of DisOrder.
   3  * Copyright (C) 2007 Richard Kettlewell
   4  *
   5  * This program is free software; you can redistribute it and/or modify
   6  * it under the terms of the GNU General Public License as published by
   7  * the Free Software Foundation; either version 2 of the License, or
   8  * (at your option) any later version.
   9  *
  10  * This program is distributed in the hope that it will be useful, but
  11  * WITHOUT ANY WARRANTY; without even the implied warranty of
  12  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  13  * General Public License for more details.
  14  *
  15  * You should have received a copy of the GNU General Public License
  16  * along with this program; if not, write to the Free Software
  17  * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
  18  * USA
  19  */
  20 /** @file clients/playrtp.c
  21  * @brief RTP player
  22  *
  23  * This player supports Linux (<a href="http://www.alsa-project.org/">ALSA</a>)
  24  * and Apple Mac (<a
  25  * href="http://developer.apple.com/audio/coreaudio.html">Core Audio</a>)
  26  * systems.  There is no support for Microsoft Windows yet, and that will in
  27  * fact probably an entirely separate program.
  28  *
  29  * The program runs (at least) three threads.  listen_thread() is responsible
  30  * for reading RTP packets off the wire and adding them to the linked list @ref
  31  * received_packets, assuming they are basically sound.  queue_thread() takes
  32  * packets off this linked list and adds them to @ref packets (an operation
  33  * which might be much slower due to contention for @ref lock).
  34  *
  35  * The main thread is responsible for actually playing audio.  In ALSA this
  36  * means it waits until ALSA says it's ready for more audio which it then
  37  * plays.  See @ref clients/playrtp-alsa.c.
  38  *
  39  * In Core Audio the main thread is only responsible for starting and stopping
  40  * play: the system does the actual playback in its own private thread, and
  41  * calls adioproc() to fetch the audio data.  See @ref
  42  * clients/playrtp-coreaudio.c.
  43  *
  44  * Sometimes it happens that there is no audio available to play.  This may
  45  * because the server went away, or a packet was dropped, or the server
  46  * deliberately did not send any sound because it encountered a silence.
  47  *
  48  * Assumptions:
  49  * - it is safe to read uint32_t values without a lock protecting them
  50  */
  51
  52 #include <config.h>
  53 #include "types.h"
  54
  55 #include <getopt.h>
  56 #include <stdio.h>
  57 #include <stdlib.h>
  58 #include <sys/socket.h>
  59 #include <sys/types.h>
  60 #include <sys/socket.h>
  61 #include <netdb.h>
  62 #include <pthread.h>
  63 #include <locale.h>
  64 #include <sys/uio.h>
  65 #include <string.h>
  66 #include <assert.h>
  67 #include <errno.h>
  68 #include <netinet/in.h>
  69 #include <sys/time.h>
  70 #include <sys/un.h>
  71 #include <unistd.h>
  72 #include <sys/mman.h>
  73 #include <fcntl.h>
  74
  75 #include "log.h"
  76 #include "mem.h"
  77 #include "configuration.h"
  78 #include "addr.h"
  79 #include "syscalls.h"
  80 #include "rtp.h"
  81 #include "defs.h"
  82 #include "vector.h"
  83 #include "heap.h"
  84 #include "timeval.h"
  85 #include "client.h"
  86 #include "playrtp.h"
  87 #include "inputline.h"
  88
  89 #define readahead linux_headers_are_borked
  90
  91 /** @brief Obsolete synonym */
  92 #ifndef IPV6_JOIN_GROUP
  93 # define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
  94 #endif
  95
  96 /** @brief RTP socket */
  97 static int rtpfd;
  98
  99 /** @brief Log output */
 100 static FILE *logfp;
 101
 102 /** @brief Output device */
 103 const char *device;
 104
 105 /** @brief Minimum low watermark
 106  *
 107  * We'll stop playing if there's only this many samples in the buffer. */
 108 unsigned minbuffer = 2 * 44100 / 10;  /* 0.2 seconds */
 109
 110 /** @brief Buffer high watermark
 111  *
 112  * We'll only start playing when this many samples are available. */
 113 static unsigned readahead = 2 * 2 * 44100;
 114
 115 /** @brief Maximum buffer size
 116  *
 117  * We'll stop reading from the network if we have this many samples. */
 118 static unsigned maxbuffer;
 119
 120 /** @brief Received packets
 121  * Protected by @ref receive_lock
 122  *
 123  * Received packets are added to this list, and queue_thread() picks them off
 124  * it and adds them to @ref packets.  Whenever a packet is added to it, @ref
 125  * receive_cond is signalled.
 126  */
 127 struct packet *received_packets;
 128
 129 /** @brief Tail of @ref received_packets
 130  * Protected by @ref receive_lock
 131  */
 132 struct packet **received_tail = &received_packets;
 133
 134 /** @brief Lock protecting @ref received_packets
 135  *
 136  * Only listen_thread() and queue_thread() ever hold this lock.  It is vital
 137  * that queue_thread() not hold it any longer than it strictly has to. */
 138 pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER;
 139
 140 /** @brief Condition variable signalled when @ref received_packets is updated
 141  *
 142  * Used by listen_thread() to notify queue_thread() that it has added another
 143  * packet to @ref received_packets. */
 144 pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER;
 145
 146 /** @brief Length of @ref received_packets */
 147 uint32_t nreceived;
 148
 149 /** @brief Binary heap of received packets */
 150 struct pheap packets;
 151
 152 /** @brief Total number of samples available
 153  *
 154  * We make this volatile because we inspect it without a protecting lock,
 155  * so the usual pthread_* guarantees aren't available.
 156  */
 157 volatile uint32_t nsamples;
 158
 159 /** @brief Timestamp of next packet to play.
 160  *
 161  * This is set to the timestamp of the last packet, plus the number of
 162  * samples it contained.  Only valid if @ref active is nonzero.
 163  */
 164 uint32_t next_timestamp;
 165
 166 /** @brief True if actively playing
 167  *
 168  * This is true when playing and false when just buffering. */
 169 int active;
 170
 171 /** @brief Lock protecting @ref packets */
 172 pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
 173
 174 /** @brief Condition variable signalled whenever @ref packets is changed */
 175 pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
 176
 177 #if HAVE_ALSA_ASOUNDLIB_H
 178 # define DEFAULT_BACKEND playrtp_alsa
 179 #elif HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
 180 # define DEFAULT_BACKEND playrtp_oss
 181 #elif HAVE_COREAUDIO_AUDIOHARDWARE_H
 182 # define DEFAULT_BACKEND playrtp_coreaudio
 183 #else
 184 # error No known backend
 185 #endif
 186
 187 /** @brief Backend to play with */
 188 static void (*backend)(void) = &DEFAULT_BACKEND;
 189
 190 HEAP_DEFINE(pheap, struct packet *, lt_packet);
 191
 192 /** @brief Control socket or NULL */
 193 const char *control_socket;
 194
 195 int16_t *dump_buffer;
 196 size_t dump_index;
 197 size_t dump_size = 44100 * 2 * 20; /* 20s */
 198
 199 static const struct option options[] = {
 200   { "help", no_argument, 0, 'h' },
 201   { "version", no_argument, 0, 'V' },
 202   { "debug", no_argument, 0, 'd' },
 203   { "device", required_argument, 0, 'D' },
 204   { "min", required_argument, 0, 'm' },
 205   { "max", required_argument, 0, 'x' },
 206   { "buffer", required_argument, 0, 'b' },
 207   { "rcvbuf", required_argument, 0, 'R' },
 208 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
 209   { "oss", no_argument, 0, 'o' },
 210 #endif
 211 #if HAVE_ALSA_ASOUNDLIB_H
 212   { "alsa", no_argument, 0, 'a' },
 213 #endif
 214 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
 215   { "core-audio", no_argument, 0, 'c' },
 216 #endif
 217   { "dump", required_argument, 0, 'r' },
 218   { "socket", required_argument, 0, 's' },
 219   { "config", required_argument, 0, 'C' },
 220   { 0, 0, 0, 0 }
 221 };
 222
 223 /** @brief Control thread
 224  *
 225  * This thread is responsible for accepting control commands from Disobedience
 226  * (or other controllers) over an AF_UNIX stream socket with a path specified
 227  * by the @c --socket option.  The protocol uses simple string commands and
 228  * replies:
 229  *
 230  * - @c stop will shut the player down
 231  * - @c query will send back the reply @c running
 232  * - anything else is ignored
 233  *
 234  * Commands and response strings terminated by shutting down the connection or
 235  * by a newline.  No attempt is made to multiplex multiple clients so it is
 236  * important that the command be sent as soon as the connection is made - it is
 237  * assumed that both parties to the protocol are entirely cooperating with one
 238  * another.
 239  */
 240 static void *control_thread(void attribute((unused)) *arg) {
 241   struct sockaddr_un sa;
 242   int sfd, cfd;
 243   char *line;
 244   socklen_t salen;
 245   FILE *fp;
 246
 247   assert(control_socket);
 248   unlink(control_socket);
 249   memset(&sa, 0, sizeof sa);
 250   sa.sun_family = AF_UNIX;
 251   strcpy(sa.sun_path, control_socket);
 252   sfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
 253   if(bind(sfd, (const struct sockaddr *)&sa, sizeof sa) < 0)
 254     fatal(errno, "error binding to %s", control_socket);
 255   if(listen(sfd, 128) < 0)
 256     fatal(errno, "error calling listen on %s", control_socket);
 257   info("listening on %s", control_socket);
 258   for(;;) {
 259     salen = sizeof sa;
 260     cfd = accept(sfd, (struct sockaddr *)&sa, &salen);
 261     if(cfd < 0) {
 262       switch(errno) {
 263       case EINTR:
 264       case EAGAIN:
 265         break;
 266       default:
 267         fatal(errno, "error calling accept on %s", control_socket);
 268       }
 269     }
 270     if(!(fp = fdopen(cfd, "r+"))) {
 271       error(errno, "error calling fdopen for %s connection", control_socket);
 272       close(cfd);
 273       continue;
 274     }
 275     if(!inputline(control_socket, fp, &line, '\n')) {
 276       if(!strcmp(line, "stop")) {
 277         info("stopped via %s", control_socket);
 278         exit(0);                          /* terminate immediately */
 279       }
 280       if(!strcmp(line, "query"))
 281         fprintf(fp, "running");
 282       xfree(line);
 283     }
 284     if(fclose(fp) < 0)
 285       error(errno, "error closing %s connection", control_socket);
 286   }
 287 }
 288
 289 /** @brief Drop the first packet
 290  *
 291  * Assumes that @ref lock is held.
 292  */
 293 static void drop_first_packet(void) {
 294   if(pheap_count(&packets)) {
 295     struct packet *const p = pheap_remove(&packets);
 296     nsamples -= p->nsamples;
 297     playrtp_free_packet(p);
 298     pthread_cond_broadcast(&cond);
 299   }
 300 }
 301
 302 /** @brief Background thread adding packets to heap
 303  *
 304  * This just transfers packets from @ref received_packets to @ref packets.  It
 305  * is important that it holds @ref receive_lock for as little time as possible,
 306  * in order to minimize the interval between calls to read() in
 307  * listen_thread().
 308  */
 309 static void *queue_thread(void attribute((unused)) *arg) {
 310   struct packet *p;
 311
 312   for(;;) {
 313     /* Get the next packet */
 314     pthread_mutex_lock(&receive_lock);
 315     while(!received_packets)
 316       pthread_cond_wait(&receive_cond, &receive_lock);
 317     p = received_packets;
 318     received_packets = p->next;
 319     if(!received_packets)
 320       received_tail = &received_packets;
 321     --nreceived;
 322     pthread_mutex_unlock(&receive_lock);
 323     /* Add it to the heap */
 324     pthread_mutex_lock(&lock);
 325     pheap_insert(&packets, p);
 326     nsamples += p->nsamples;
 327     pthread_cond_broadcast(&cond);
 328     pthread_mutex_unlock(&lock);
 329   }
 330 }
 331
 332 /** @brief Background thread collecting samples
 333  *
 334  * This function collects samples, perhaps converts them to the target format,
 335  * and adds them to the packet list.
 336  *
 337  * It is crucial that the gap between successive calls to read() is as small as
 338  * possible: otherwise packets will be dropped.
 339  *
 340  * We use a binary heap to ensure that the unavoidable effort is at worst
 341  * logarithmic in the total number of packets - in fact if packets are mostly
 342  * received in order then we will largely do constant work per packet since the
 343  * newest packet will always be last.
 344  *
 345  * Of more concern is that we must acquire the lock on the heap to add a packet
 346  * to it.  If this proves a problem in practice then the answer would be
 347  * (probably doubly) linked list with new packets added the end and a second
 348  * thread which reads packets off the list and adds them to the heap.
 349  *
 350  * We keep memory allocation (mostly) very fast by keeping pre-allocated
 351  * packets around; see @ref playrtp_new_packet().
 352  */
 353 static void *listen_thread(void attribute((unused)) *arg) {
 354   struct packet *p = 0;
 355   int n;
 356   struct rtp_header header;
 357   uint16_t seq;
 358   uint32_t timestamp;
 359   struct iovec iov[2];
 360
 361   for(;;) {
 362     if(!p)
 363       p = playrtp_new_packet();
 364     iov[0].iov_base = &header;
 365     iov[0].iov_len = sizeof header;
 366     iov[1].iov_base = p->samples_raw;
 367     iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
 368     n = readv(rtpfd, iov, 2);
 369     if(n < 0) {
 370       switch(errno) {
 371       case EINTR:
 372         continue;
 373       default:
 374         fatal(errno, "error reading from socket");
 375       }
 376     }
 377     /* Ignore too-short packets */
 378     if((size_t)n <= sizeof (struct rtp_header)) {
 379       info("ignored a short packet");
 380       continue;
 381     }
 382     timestamp = htonl(header.timestamp);
 383     seq = htons(header.seq);
 384     /* Ignore packets in the past */
 385     if(active && lt(timestamp, next_timestamp)) {
 386       info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
 387            timestamp, next_timestamp);
 388       continue;
 389     }
 390     p->next = 0;
 391     p->flags = 0;
 392     p->timestamp = timestamp;
 393     /* Convert to target format */
 394     if(header.mpt & 0x80)
 395       p->flags |= IDLE;
 396     switch(header.mpt & 0x7F) {
 397     case 10:
 398       p->nsamples = (n - sizeof header) / sizeof(uint16_t);
 399       break;
 400       /* TODO support other RFC3551 media types (when the speaker does) */
 401     default:
 402       fatal(0, "unsupported RTP payload type %d",
 403             header.mpt & 0x7F);
 404     }
 405     if(logfp)
 406       fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
 407               seq, timestamp, p->nsamples, timestamp + p->nsamples);
 408     /* Stop reading if we've reached the maximum.
 409      *
 410      * This is rather unsatisfactory: it means that if packets get heavily
 411      * out of order then we guarantee dropouts.  But for now... */
 412     if(nsamples >= maxbuffer) {
 413       pthread_mutex_lock(&lock);
 414       while(nsamples >= maxbuffer)
 415         pthread_cond_wait(&cond, &lock);
 416       pthread_mutex_unlock(&lock);
 417     }
 418     /* Add the packet to the receive queue */
 419     pthread_mutex_lock(&receive_lock);
 420     *received_tail = p;
 421     received_tail = &p->next;
 422     ++nreceived;
 423     pthread_cond_signal(&receive_cond);
 424     pthread_mutex_unlock(&receive_lock);
 425     /* We'll need a new packet */
 426     p = 0;
 427   }
 428 }
 429
 430 /** @brief Wait until the buffer is adequately full
 431  *
 432  * Must be called with @ref lock held.
 433  */
 434 void playrtp_fill_buffer(void) {
 435   while(nsamples)
 436     drop_first_packet();
 437   info("Buffering...");
 438   while(nsamples < readahead)
 439     pthread_cond_wait(&cond, &lock);
 440   next_timestamp = pheap_first(&packets)->timestamp;
 441   active = 1;
 442 }
 443
 444 /** @brief Find next packet
 445  * @return Packet to play or NULL if none found
 446  *
 447  * The return packet is merely guaranteed not to be in the past: it might be
 448  * the first packet in the future rather than one that is actually suitable to
 449  * play.
 450  *
 451  * Must be called with @ref lock held.
 452  */
 453 struct packet *playrtp_next_packet(void) {
 454   while(pheap_count(&packets)) {
 455     struct packet *const p = pheap_first(&packets);
 456     if(le(p->timestamp + p->nsamples, next_timestamp)) {
 457       /* This packet is in the past.  Drop it and try another one. */
 458       drop_first_packet();
 459     } else
 460       /* This packet is NOT in the past.  (It might be in the future
 461        * however.) */
 462       return p;
 463   }
 464   return 0;
 465 }
 466
 467 /** @brief Play an RTP stream
 468  *
 469  * This is the guts of the program.  It is responsible for:
 470  * - starting the listening thread
 471  * - opening the audio device
 472  * - reading ahead to build up a buffer
 473  * - arranging for audio to be played
 474  * - detecting when the buffer has got too small and re-buffering
 475  */
 476 static void play_rtp(void) {
 477   pthread_t ltid;
 478   int err;
 479
 480   /* We receive and convert audio data in a background thread */
 481   if((err = pthread_create(&ltid, 0, listen_thread, 0)))
 482     fatal(err, "pthread_create listen_thread");
 483   /* We have a second thread to add received packets to the queue */
 484   if((err = pthread_create(&ltid, 0, queue_thread, 0)))
 485     fatal(err, "pthread_create queue_thread");
 486   /* The rest of the work is backend-specific */
 487   backend();
 488 }
 489
 490 /* display usage message and terminate */
 491 static void help(void) {
 492   xprintf("Usage:\n"
 493           "  disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
 494           "Options:\n"
 495           "  --device, -D DEVICE     Output device\n"
 496           "  --min, -m FRAMES        Buffer low water mark\n"
 497           "  --buffer, -b FRAMES     Buffer high water mark\n"
 498           "  --max, -x FRAMES        Buffer maximum size\n"
 499           "  --rcvbuf, -R BYTES      Socket receive buffer size\n"
 500           "  --config, -C PATH       Set configuration file\n"
 501 #if HAVE_ALSA_ASOUNDLIB_H
 502           "  --alsa, -a              Use ALSA to play audio\n"
 503 #endif
 504 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
 505           "  --oss, -o               Use OSS to play audio\n"
 506 #endif
 507 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
 508           "  --core-audio, -c        Use Core Audio to play audio\n"
 509 #endif
 510           "  --help, -h              Display usage message\n"
 511           "  --version, -V           Display version number\n"
 512           );
 513   xfclose(stdout);
 514   exit(0);
 515 }
 516
 517 /* display version number and terminate */
 518 static void version(void) {
 519   xprintf("disorder-playrtp version %s\n", disorder_version_string);
 520   xfclose(stdout);
 521   exit(0);
 522 }
 523
 524 int main(int argc, char **argv) {
 525   int n, err;
 526   struct addrinfo *res;
 527   struct stringlist sl;
 528   char *sockname;
 529   int rcvbuf, target_rcvbuf = 131072;
 530   socklen_t len;
 531   struct ip_mreq mreq;
 532   struct ipv6_mreq mreq6;
 533   disorder_client *c;
 534   char *address, *port;
 535   int is_multicast;
 536   union any_sockaddr {
 537     struct sockaddr sa;
 538     struct sockaddr_in in;
 539     struct sockaddr_in6 in6;
 540   };
 541   union any_sockaddr mgroup;
 542   const char *dumpfile = 0;
 543
 544   static const struct addrinfo prefs = {
 545     AI_PASSIVE,
 546     PF_INET,
 547     SOCK_DGRAM,
 548     IPPROTO_UDP,
 549     0,
 550     0,
 551     0,
 552     0
 553   };
 554
 555   mem_init();
 556   if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
 557   while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aocC:r", options, 0)) >= 0) {
 558     switch(n) {
 559     case 'h': help();
 560     case 'V': version();
 561     case 'd': debugging = 1; break;
 562     case 'D': device = optarg; break;
 563     case 'm': minbuffer = 2 * atol(optarg); break;
 564     case 'b': readahead = 2 * atol(optarg); break;
 565     case 'x': maxbuffer = 2 * atol(optarg); break;
 566     case 'L': logfp = fopen(optarg, "w"); break;
 567     case 'R': target_rcvbuf = atoi(optarg); break;
 568 #if HAVE_ALSA_ASOUNDLIB_H
 569     case 'a': backend = playrtp_alsa; break;
 570 #endif
 571 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
 572     case 'o': backend = playrtp_oss; break;
 573 #endif
 574 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
 575     case 'c': backend = playrtp_coreaudio; break;
 576 #endif
 577     case 'C': configfile = optarg; break;
 578     case 's': control_socket = optarg; break;
 579     case 'r': dumpfile = optarg; break;
 580     default: fatal(0, "invalid option");
 581     }
 582   }
 583   if(config_read(0)) fatal(0, "cannot read configuration");
 584   if(!maxbuffer)
 585     maxbuffer = 4 * readahead;
 586   argc -= optind;
 587   argv += optind;
 588   switch(argc) {
 589   case 0:
 590     /* Get configuration from server */
 591     if(!(c = disorder_new(1))) exit(EXIT_FAILURE);
 592     if(disorder_connect(c)) exit(EXIT_FAILURE);
 593     if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE);
 594     sl.n = 2;
 595     sl.s = xcalloc(2, sizeof *sl.s);
 596     sl.s[0] = address;
 597     sl.s[1] = port;
 598     break;
 599   case 1:
 600   case 2:
 601     /* Use command-line ADDRESS+PORT or just PORT */
 602     sl.n = argc;
 603     sl.s = argv;
 604     break;
 605   default:
 606     fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
 607   }
 608   /* Look up address and port */
 609   if(!(res = get_address(&sl, &prefs, &sockname)))
 610     exit(1);
 611   /* Create the socket */
 612   if((rtpfd = socket(res->ai_family,
 613                      res->ai_socktype,
 614                      res->ai_protocol)) < 0)
 615     fatal(errno, "error creating socket");
 616   /* Stash the multicast group address */
 617   if((is_multicast = multicast(res->ai_addr))) {
 618     memcpy(&mgroup, res->ai_addr, res->ai_addrlen);
 619     switch(res->ai_addr->sa_family) {
 620     case AF_INET:
 621       mgroup.in.sin_port = 0;
 622       break;
 623     case AF_INET6:
 624       mgroup.in6.sin6_port = 0;
 625       break;
 626     }
 627   }
 628   /* Bind to 0/port */
 629   switch(res->ai_addr->sa_family) {
 630   case AF_INET:
 631     memset(&((struct sockaddr_in *)res->ai_addr)->sin_addr, 0,
 632            sizeof (struct in_addr));
 633     break;
 634   case AF_INET6:
 635     memset(&((struct sockaddr_in6 *)res->ai_addr)->sin6_addr, 0,
 636            sizeof (struct in6_addr));
 637     break;
 638   default:
 639     fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family);
 640   }
 641   if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
 642     fatal(errno, "error binding socket to %s", sockname);
 643   if(is_multicast) {
 644     switch(mgroup.sa.sa_family) {
 645     case PF_INET:
 646       mreq.imr_multiaddr = mgroup.in.sin_addr;
 647       mreq.imr_interface.s_addr = 0;      /* use primary interface */
 648       if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP,
 649                     &mreq, sizeof mreq) < 0)
 650         fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP");
 651       break;
 652     case PF_INET6:
 653       mreq6.ipv6mr_multiaddr = mgroup.in6.sin6_addr;
 654       memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface);
 655       if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP,
 656                     &mreq6, sizeof mreq6) < 0)
 657         fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP");
 658       break;
 659     default:
 660       fatal(0, "unsupported address family %d", res->ai_family);
 661     }
 662     info("listening on %s multicast group %s",
 663          format_sockaddr(res->ai_addr), format_sockaddr(&mgroup.sa));
 664   } else
 665     info("listening on %s", format_sockaddr(res->ai_addr));
 666   len = sizeof rcvbuf;
 667   if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
 668     fatal(errno, "error calling getsockopt SO_RCVBUF");
 669   if(target_rcvbuf > rcvbuf) {
 670     if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
 671                   &target_rcvbuf, sizeof target_rcvbuf) < 0)
 672       error(errno, "error calling setsockopt SO_RCVBUF %d",
 673             target_rcvbuf);
 674       /* We try to carry on anyway */
 675     else
 676       info("changed socket receive buffer from %d to %d",
 677            rcvbuf, target_rcvbuf);
 678   } else
 679     info("default socket receive buffer %d", rcvbuf);
 680   if(logfp)
 681     info("WARNING: -L option can impact performance");
 682   if(control_socket) {
 683     pthread_t tid;
 684
 685     if((err = pthread_create(&tid, 0, control_thread, 0)))
 686       fatal(err, "pthread_create control_thread");
 687   }
 688   if(dumpfile) {
 689     int fd;
 690     unsigned char buffer[65536];
 691     size_t written;
 692
 693     if((fd = open(dumpfile, O_RDWR|O_TRUNC|O_CREAT, 0666)) < 0)
 694       fatal(errno, "opening %s", dumpfile);
 695     /* Fill with 0s to a suitable size */
 696     memset(buffer, 0, sizeof buffer);
 697     for(written = 0; written < dump_size * sizeof(int16_t);
 698         written += sizeof buffer) {
 699       if(write(fd, buffer, sizeof buffer) < 0)
 700         fatal(errno, "clearing %s", dumpfile);
 701     }
 702     /* Map the buffer into memory for convenience */
 703     dump_buffer = mmap(0, dump_size * sizeof(int16_t), PROT_READ|PROT_WRITE,
 704                        MAP_SHARED, fd, 0);
 705     if(dump_buffer == (void *)-1)
 706       fatal(errno, "mapping %s", dumpfile);
 707     info("dumping to %s", dumpfile);
 708   }
 709   play_rtp();
 710   return 0;
 711 }
 712
 713 /*
 714 Local Variables:
 715 c-basic-offset:2
 716 comment-column:40
 717 fill-column:79
 718 indent-tabs-mode:nil
 719 End:
 720 */